[..] |
ASTERISK-01000: [patch] asterisk-sounds/Makefile missing target for silence |
ASTERISK-01001: [patch] Recognise distinctive ring when presented after CID |
ASTERISK-01002: request wav moh type - mpg123 cause crash |
ASTERISK-01003: SIP channels for FWD stale after ~10 minutes of a reload |
ASTERISK-01004: [patch] memory leak when using mysql_friends |
ASTERISK-01005: [patch] memory leak when using mysql_friends |
ASTERISK-01006: Invalid Q.931 setup information for data call |
ASTERISK-01007: [patch] configs/sip.conf additions |
ASTERISK-01008: [patch] Change error text for maximum call group |
ASTERISK-01009: Supervised transfers with SIP REFER/Replaces do not work with Sipura and X-Pro |
ASTERISK-01010: Answering call on h.323 causes Asterisk to crash |
ASTERISK-01011: Switch statement fails to process 3 digit dialplan |
ASTERISK-01012: pbx_spool |
ASTERISK-01013: [patch] Binding RTP to a specific interface |
ASTERISK-01014: long wait time until RING status |
ASTERISK-01015: chan_h323 will dumps core with CLI reload |
ASTERISK-01016: [patch] RemoveQueueMember & AddQueueMember problems |
ASTERISK-01017: Calls dropped with no known reason |
ASTERISK-01018: Incomplete RFC2833 support |
ASTERISK-01019: [patch] add doublehash transfer option to parking.conf |
ASTERISK-01020: [patch] Ability to do PINs with dynamic conferences |
ASTERISK-01021: [patch] Debugging output when parsing SDP codecs |
ASTERISK-01022: [request] more CLI dialplan methods |
ASTERISK-01023: Serious bug in 'ast_rtp_raw_write' |
ASTERISK-01024: CDR mySQL records |
ASTERISK-01025: SIP inUse counter fails to decrease |
ASTERISK-01026: REGISTER branch number not being incremented/changed after transaction |
ASTERISK-01027: SIP DTMF INFO mode does not work reliably |
ASTERISK-01028: [PATCH] fix multiple detection of rtp dtmf events with eg cisco phones |
ASTERISK-01029: Sample iax.conf has typo in maxexcessbuffer |
ASTERISK-01030: Autocreatepeer can't be billed in sip call. |
ASTERISK-01031: typo in chan_sip.c |
ASTERISK-01032: [post-1.0][patch] IAX2 not passing caller*id presentation indicator / ast_channel restrictcid-flag not set |
ASTERISK-01033: AgentLogin, Hard hangup on a call from the queue, freezes the SIP channel the agent is logged in on |
ASTERISK-01034: Asterisk crash when call initiated from X-Pro (SIP) to cisco 7940(SIP) and conference bridge with 7910 (skinny) used |
ASTERISK-01035: Caller ID info does not distinguish between international and national |
ASTERISK-01036: [workaround?] wcfxo and wcfxs Segmentation Fault in kernel 2.6.2/2.6.3 (related to CONFIG_PREEMPT) |
ASTERISK-01037: "codec0 is not codec1" bug in ast_rtp_bridge |
ASTERISK-01038: Logger Problem |
ASTERISK-01039: compile errors on openbsd (from dev list) |
ASTERISK-01040: [patch] Only one phone in queue rings at a time |
ASTERISK-01041: [patch] video format description bug into an INVITE message |
ASTERISK-01042: ztmonitor no longer works |
ASTERISK-01043: Secondary Zap Channel Zap/X-2 gets locked in weird state |
ASTERISK-01044: action: status results in short listing on loaded box |
ASTERISK-01045: Queue / Agent timeout logic issues |
ASTERISK-01046: [patch] mysql-friends times out after 120 seconds. Also memory leak. |
ASTERISK-01047: [patch] SIP Info problems may cause call teardown |
ASTERISK-01048: WARNING[3065886]: chan_iax2.c:2512 iax2_send: Out of trunk data space on call number 16389, dropping |
ASTERISK-01049: Dial with SIP and IAX target not indicating ring to caller |
ASTERISK-01050: [workaround] Queue does not work when called from the 't' extension |
ASTERISK-01051: patches to makefiles for cross-compliation |
ASTERISK-01052: 2.6 makefile problems |
ASTERISK-01053: two more typos in chan_sip.c |
ASTERISK-01054: REGISTER statement in sip.conf - wrong extensions |
ASTERISK-01055: PRI error and warnings |
ASTERISK-01056: [patch] Add a variable to hold the filename created in the Record application |
ASTERISK-01057: SIP INVITE and To: missing username when using mysql and sipfriends table |
ASTERISK-01058: incominglimit not working |
ASTERISK-01059: Asterisk segfault on extensions.conf when reloading |
ASTERISK-01060: [patch] Fedora prefers /etc/modprobe.conf to /etc/modules.conf |
ASTERISK-01061: [patch] Numbers are always read in english |
ASTERISK-01062: [src audit] replacement for strncpy() function used everywhere in asterisk |
ASTERISK-01063: Changing password on a voicemail box changes it on all same-numbered boxes on all contexts |
ASTERISK-01064: [patch] Allow the Playback application to play the first file it finds from a list |
ASTERISK-01065: [patch] Configure usage of CLASS features on chan_zap |
ASTERISK-01066: [te410p] No interrupts after warm restart of host machine |
ASTERISK-01067: touch-tone used to terminate recording is heard on playback |
ASTERISK-01068: [patch] Allow calls to/from unknown destinations (SRV) maintain callerid |
ASTERISK-01069: CALLERIDNUM strips the dots |
ASTERISK-01070: chan_local cannot be used for making SIP SRV based calls. |
ASTERISK-01071: System call in dialplan erroneously reports failure |
ASTERISK-01072: Provide the current language in a variable |
ASTERISK-01073: [patch] zonedata for Greece |
ASTERISK-01074: Premature CONNECT while overlap dialing |
ASTERISK-01075: [bounty] Need to have 2B channel transfer on PRI working |
ASTERISK-01076: [patch][post 1.4] Add Pre Acknowledgement Message to AgentCallbackLogin |
ASTERISK-01077: Minor language correction |
ASTERISK-01078: [patch] app_cut corruption |
ASTERISK-01079: safe_asterisk not always working as it should |
ASTERISK-01080: [patch] Adds nat, reinvite, accountcode, amaflags, callerid, restrictcid to mySQL functionality |
ASTERISK-01081: WAV49 file recorded can not be played |
ASTERISK-01082: [patch] MailboxExists |
ASTERISK-01083: Return before spin_unlock - found in code review |
ASTERISK-01084: [request] let codec negotiation more flexible |
ASTERISK-01085: wct4xxp does not set alarm if channels are not open |
ASTERISK-01086: Asterisk crashes on 7960 -> * -> iax2 -> * -> TE410P when using GSM |
ASTERISK-01087: SIP/IAX2 show channels: delay and jitter wrong output |
ASTERISK-01088: ${HANGUPCAUSE} still returns 0 |
ASTERISK-01089: [request] Allow ADSI to be configured on skinny channels |
ASTERISK-01090: [request] 911 call handling |
ASTERISK-01091: Digits in other languages |
ASTERISK-01092: wrong message for vm-msginstruct.gsm |
ASTERISK-01093: chan_h323 compatibility with Cisco CallManager |
ASTERISK-01094: [patch] manager event flow control and write patch |
ASTERISK-01095: [patch] ast_moh_stop called before bridge |
ASTERISK-01096: duration= written to new file if calling channel hangs up after message deletion |
ASTERISK-01097: At a when call made to voicemail with alaw & ulaw slowed down prompt playback is audible |
ASTERISK-01098: G.726 format |
ASTERISK-01099: CSeq sip header field |
ASTERISK-01100: ringringbug ? |
ASTERISK-01101: AbsoluteTimeout takes into account ringing time |
ASTERISK-01102: chan_alsa.so aborts |
ASTERISK-01103: Gastman still segfaulting |
ASTERISK-01104: asterisk -xr not showing full response. |
ASTERISK-01105: IAX possible problem |
ASTERISK-01106: /proc/zaptel/1 size limit |
ASTERISK-01107: asterisk command line options need to be in correct order. |
ASTERISK-01108: [patch] chan_mgcp features / fixes combo |
ASTERISK-01109: show channels shows weird extension for Zaptel channels. |
ASTERISK-01110: [patch] callerid number not hidden when restrictcid is on |
ASTERISK-01111: zaptel 0.8.1 not compile with 2.6 |
ASTERISK-01112: App MeetMe Features |
ASTERISK-01113: First call after MGCP endpoint restart fails |
ASTERISK-01114: IETF Draft: SIP Extensions for Caller Identity and Privacy. |
ASTERISK-01115: chan_iax2 (cvs) doesn't compile on FreeBSD |
ASTERISK-01116: [patch] typo in mysql_update_peer , Tag v1-0_stable |
ASTERISK-01117: [patch] Voicemail answers the channel regardless of existance of mailbox |
ASTERISK-01118: crash core dump, signal 11, libc |
ASTERISK-01119: [patch] Enhance sample.conf files to prevent common questions/errors |
ASTERISK-01120: v1-0_stable quits with -v and no console |
ASTERISK-01121: TDM PCI Master abort |
ASTERISK-01122: [patch] integrated moh supporting both mp3 and ogg |
ASTERISK-01123: [request] SIP authentication with HTTP_DIGEST QOP values |
ASTERISK-01124: [patch] ChanIsAvailable gets into a loop when having wrong args |
ASTERISK-01125: absence of cdr-csv directory gives no warning |
ASTERISK-01126: Probable error in string handling in Cut function or DB storage |
ASTERISK-01127: [patch] Timestamps on console |
ASTERISK-01128: ParkedCall Event Generates Spurious Newline in Mgr Protocol |
ASTERISK-01129: ParkedCalls Getting Dropped |
ASTERISK-01130: MusicOnHold Interactions with Parked Calls |
ASTERISK-01131: enumlookup goes to priority+1 on parse error |
ASTERISK-01132: [patch] Remove #1084 hack and fix the real bug! |
ASTERISK-01133: Crash in ChanIsAvail when checking for a lot of users |
ASTERISK-01134: [patch] digittimeout added for app_dial |
ASTERISK-01135: Asterisk do not record DST correctly when using macros |
ASTERISK-01136: ncurses and ncurses-devel are build requirements also |
ASTERISK-01137: ncurses and ncurses-devel are build requirements also |
ASTERISK-01138: ncurses and ncurses-devel are build requirements also |
ASTERISK-01139: ncurses and ncurses-devel are build requirements also |
ASTERISK-01140: bison is a build requirement and should be listed on the download section of the Asterisk website. |
ASTERISK-01141: zlib-devel is a build requirement and should be listed on the download section of the Asterisk website. |
ASTERISK-01142: cli.c leaking memory |
ASTERISK-01143: Old code in dns.c breaks enum lookups |
ASTERISK-01144: Asterisk crashes when a call originated from an AGI is parked |
ASTERISK-01145: Wrong hour in GotoIfTime locks channels |
ASTERISK-01146: [patch] app_voicemail strncpy() auditing |
ASTERISK-01147: chan_h323.so: undefined symbol: _ZTI19H323AudioCapability |
ASTERISK-01148: Build Instructions on www.asterisk.org |
ASTERISK-01149: MGCP transfer procedures not compliant |
ASTERISK-01150: Microphone doesn't work with ALSA |
ASTERISK-01151: PostgreSQL addition to chan_iax.c and chan_iax2.c |
ASTERISK-01152: PostgreSQL addition to chan_sip.c |
ASTERISK-01153: [patch] PostgreSQL addition to chan_sip.c |
ASTERISK-01154: Roaming Extensions |
ASTERISK-01155: pri->pvt[chan]->owner deferenced when NULL during PRI hangup |
ASTERISK-01156: [patch] Queue membership (agents, NOT callers) should generate manager events |
ASTERISK-01157: Please remove ASTERISK@metrotel.net from the Asterisk-Dev list |
ASTERISK-01158: Alter IAX2 protocol to prevent competing registrations from multiple devices |
ASTERISK-01159: 'p' and 'd' flags mutually exclusive with x100p but not ztdummy |
ASTERISK-01160: Unable to Park Calls using Native SIP Transfer |
ASTERISK-01161: Compile error in chan_iax2.c |
ASTERISK-01162: usedistinctiveringdetection placed anywhere in zapata.conf is active on all channels |
ASTERISK-01163: Verbosity problem |
ASTERISK-01164: a control indication is missing. |
ASTERISK-01165: [patch] ast_context_add_switch2 fix for allowing multiple switches from one context |
ASTERISK-01166: zapata.conf not reloaded on asterisk reload. |
ASTERISK-01167: mgcp segfault |
ASTERISK-01168: auth=md5 doesnt work on amd64 |
ASTERISK-01169: asterisk in deadlock if enable overlap in the span |
ASTERISK-01170: [patch-sortof] SIP users blocked in ACL can still place calls |
ASTERISK-01171: IAX.CONF parse error |
ASTERISK-01172: iax.conf and reload do not play well together |
ASTERISK-01173: [patch] 'show voicemail users' doesn't work with USE_MYSQL_VM_INTERFACE=1 |
ASTERISK-01174: Hookflash from Voicemail to pick up incoming call weirdness |
ASTERISK-01175: No digest authentication reply to 401 - Unauthorized |
ASTERISK-01176: [patch] Eval |
ASTERISK-01177: Zapscan coredump |
ASTERISK-01178: Manager originate ignores Exten parameter. |
ASTERISK-01179: [patch] Host isn't dynamic error message - changed. |
ASTERISK-01180: pulse dial endless loop deadlock |
ASTERISK-01181: [post-1.0] [patch] Modify mailbox path on filesystem to allow for MANY boxes per context |
ASTERISK-01182: 302 "Moved Temporarily" clears Caller*ID |
ASTERISK-01183: [patch] extensions.conf.sample addition 'Z' matching |
ASTERISK-01184: Disconecting when running agi script crashes Asterisk when running in daemon mode |
ASTERISK-01185: Caller ID Name is not Available via CALLERIDNAME variable... |
ASTERISK-01186: [patch] RADIUS Authentication Accounting and call routing |
ASTERISK-01187: When parking a call system seg faults... |
ASTERISK-01188: Timestamp Problems on the RTP Stream |
ASTERISK-01189: iax2 to te410p... how is this possible.. ? |
ASTERISK-01190: Stretched Audio on Voicemail |
ASTERISK-01191: [patch] Exec |
ASTERISK-01192: [patch]Memory leak in handle_recordfile |
ASTERISK-01193: [post-1.0][patch] support to set via module param max loop current on the proslic |
ASTERISK-01194: [patch] Voicemail says goodbye twice in a row if timeout from main menu |
ASTERISK-01195: Potential lockup when read() from a pipe to mpg123 causes channel to become unusable and pbx turns call away |
ASTERISK-01196: Patches to make * compile on OpenBSD |
ASTERISK-01197: execute command |
ASTERISK-01198: IAXy with ADPCM (firmware v12) malfunction on incoming calls |
ASTERISK-01199: One way audio with IAXy (version 12) |
ASTERISK-01200: Zaptel driver memory management debug warning |
ASTERISK-01201: [patch] Voicemail with mysql option on does not allow same extension in multiple contexts |
ASTERISK-01202: [request] go to fax extension when fax signal received during voice call without user interaction |
ASTERISK-01203: [PATCH] incorrect endpos when silence detection used during record |
ASTERISK-01204: how to use Originate Variable in Manager API |
ASTERISK-01205: Cisco ATA does not recieve sound from IAXy |
ASTERISK-01206: Invalid Protocol Implementation - Very easy fix |
ASTERISK-01207: show uptime giving weird results... |
ASTERISK-01208: debug logs do not write |
ASTERISK-01209: Manager interface and -rx gives short list. |
ASTERISK-01210: When loading * and unable to load chan_iax2 (and iax1) it segment fault |
ASTERISK-01211: Codec negotiation with re-invites |
ASTERISK-01212: G.729 codec leaves shared memory segments |
ASTERISK-01213: Very Bad Audio (Jitter) Sip-to-Sip |
ASTERISK-01214: [patch] Incoming H.323 calls via E164 or E164 Prefixes get sent to configured context |
ASTERISK-01215: Voicetronix channel driver: chan_vpb.c failes to compile |
ASTERISK-01216: Internal call to queue drops after exactly 1 minute |
ASTERISK-01217: CVS head and 1.0 block every 30 minutes ( since >=22feb) |
ASTERISK-01218: a ringingtimeout for calls ( iax, h323,sip) |
ASTERISK-01219: 0.7.X now Strips out Private CID's |
ASTERISK-01220: [feature request] for MGCP and RADVISION STACK |
ASTERISK-01221: [patch] ${LEN()} crashes on string containing colon (:) |
ASTERISK-01222: [patch] New API for modularizing external queries outside of Asterisk Base |
ASTERISK-01223: [BOUNTY] NOTICE for dtmfmode is inband and codec is not ulaw or alaw |
ASTERISK-01224: BOUNTY $50 to add dnid and rdnis support to chan_sip |
ASTERISK-01225: [patch] Impliments the inbound part of sip-notify(a proprietary cisco format) |
ASTERISK-01226: Passing original timestamps requires validation of ALL sources |
ASTERISK-01227: [patch] Discontinous RTP transmission (DTX) support |
ASTERISK-01228: Q931 provider throws error on restart Command |
ASTERISK-01229: chan_sip.c: LAN traffic + using sip secret causes registration to fail/drop/no-authorization |
ASTERISK-01230: chan_sip.c : using g729 with * and snom 200 results in drops call after 6 secs : max_retries_exceeded |
ASTERISK-01231: md5/secret not possible with Draytek 2600V: chan_sip.c |
ASTERISK-01232: Draytek 2600V when placing SIP calls to PSTN user on analogue line keeps call open after Dtek hangup so PSTN user can't call out |
ASTERISK-01233: segmentation fault |
ASTERISK-01234: IAX taking down linked boxes |
ASTERISK-01235: Dial Broken, Timeout == 0 & Immediate Bridge? |
ASTERISK-01236: chan_sip.c : using g729 with * and snom 200 results in dropped call 6 secs after it is answered : error is max_retries_exceeded |
ASTERISK-01237: app_queue option "n" latches |
ASTERISK-01238: app_system not working under fedora FC1 |
ASTERISK-01239: [request] app_queue needs more options |
ASTERISK-01240: [patch] Minimum length of a voicemail |
ASTERISK-01241: [request] add ackcallkey to agents.conf |
ASTERISK-01242: Calling a queue on a remote asterisk server results in silence |
ASTERISK-01243: zapscan triggers segfault when monitoring a chan that has no type set |
ASTERISK-01244: After an unknown event all outbound calls seize to hang up cleanly |
ASTERISK-01245: One-way audio with chan_capi -> sip |
ASTERISK-01246: [patch] IAX2 show users - keys |
ASTERISK-01247: asterisk coredump when doing multiple simultaneous ilbc/SIP -> gsm/IAX2 transcodings (latest cvs) |
ASTERISK-01248: [patch] Change 'show locals' to "local show channels" |
ASTERISK-01249: Record entire conversation |
ASTERISK-01250: Dial App is not working when trying to forward to an external Number |
ASTERISK-01251: segentation fault with qualify=yes |
ASTERISK-01252: app_Read does not always work |
ASTERISK-01253: TimeStamp problem in new rtp.c file |
ASTERISK-01254: Enhance app_read to accept a maximum of N digits |
ASTERISK-01255: app_voicemail fails to release channel if mailapp hangs |
ASTERISK-01256: [patch] SIP disabled is LOG_ERROR |
ASTERISK-01257: When I use the board TE410P I can't make outgoing call to telecom provider |
ASTERISK-01258: DTMF CLIP not supported by asterisk |
ASTERISK-01259: asterisk will native bridged trunked iax to non trunked iax which doesn't work |
ASTERISK-01260: manager and outgoing calls via CAPI are not cancelled if immediately hung up |
ASTERISK-01261: [patch] Update monitor app to exec custom script when channel closes |
ASTERISK-01262: [patch] AddQueueMember/RemoveQueueMember: improved support for dynamic extension determination |
ASTERISK-01263: [patch] Custom CLI prompt |
ASTERISK-01264: [patch]Document 'o' extension in voicemail app |
ASTERISK-01265: [patch] Misc stability fixes |
ASTERISK-01266: codec_g729b.so needs to be rebuilt to handle new frame format |
ASTERISK-01267: Using SIPFRIENDS, segmentation fault if SIP peer does not exist |
ASTERISK-01268: ast_translator() bug? |
ASTERISK-01269: G.723.1 samples field value is wrong |
ASTERISK-01270: Asterisk Coredumps |
ASTERISK-01271: directory command in extensions.conf fails |
ASTERISK-01272: Asterisk Crash when given Reload when a Call is in Progress |
ASTERISK-01273: command line completion problems |
ASTERISK-01274: [patch] voicemail calculates duration incorrectly and does not delete txtfile if duration<minduration |
ASTERISK-01275: coredump with latest cvs |
ASTERISK-01276: CVS-HEAD rtp issues cause choppy audio! We are working on it! :) |
ASTERISK-01277: [request] Voicemail Rewind Functionality |
ASTERISK-01278: call in outgoing is placing call but after pickup hear dtmf pulsed digits |
ASTERISK-01279: cache not working |
ASTERISK-01280: TE410P Interrupts Freese the Box |
ASTERISK-01281: [patch] transaction id made unsigned + delete stale connections |
ASTERISK-01282: No DTMF before SIP CONNECT |
ASTERISK-01283: [patch] configure emailsubject= in same manner of emailbody= |
ASTERISK-01284: [patch] ast_expr can't handle quoted strings, bad error messages, multiple spaces... |
ASTERISK-01285: Double link events while linking two channels |
ASTERISK-01286: CallParking - Double Parked Conersation crashs * |
ASTERISK-01287: [patch] Outcalling from within meetme |
ASTERISK-01288: zapata.conf needs to be updated for mwi & mailbox contexts |
ASTERISK-01289: Manager event for call setup/tear down |
ASTERISK-01290: [patch] using select() with alsa console drivers is unreliable, unless snd_pcm_poll_descriptors_revents() is called after it. |
ASTERISK-01291: [patch] set/append CDR User Field for manager interface |
ASTERISK-01292: [patch] Document the maximum length of voicemail email message body |
ASTERISK-01293: [patch]Application to send event to manager clients |
ASTERISK-01294: Loss of the agi-name when calling an exec application. |
ASTERISK-01295: registering new g729 key causes failure with safe_asterisk |
ASTERISK-01296: Mgcp used to stop functioning before we applied the following |
ASTERISK-01297: [patch] Documentation for ASTERISK_PROMPT |
ASTERISK-01298: libpri does not support "SDN Marking" |
ASTERISK-01299: libpri does not support AT&T feature "Dialed Number Preferred" |
ASTERISK-01300: Retry problem in pbx_spool module |
ASTERISK-01301: [workaround] Documentation of agentcallback problems |
ASTERISK-01302: [block] chan_h323 consumes all available CPU |
ASTERISK-01303: zttool show incorrect call counts |
ASTERISK-01304: Perform proper rounding in calc_txstamp |
ASTERISK-01305: Bad branch id in VIA |
ASTERISK-01306: [bounty] MGCP Media Gateway support (Asterisk as Client to Call Agent) |
ASTERISK-01307: Directory cmd & voicemail-mysql |
ASTERISK-01308: Using one SIP account and login from several locations |
ASTERISK-01309: Error's Compiling all code under FreeBSD 5.2.1 |
ASTERISK-01310: [patch] More gastman fixups |
ASTERISK-01311: [PATCH] Repaired building for RedHat 6.2 |
ASTERISK-01312: FXO_KS signalled Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State |
ASTERISK-01313: OSS/ALSA Channels fail to release /dev/dsp after first call recieved on console. |
ASTERISK-01314: [REQUEST] Bug Marshal Test Server being built! |
ASTERISK-01315: Zaptel module makes system unstable |
ASTERISK-01316: [patch] rtp.c does not allow to bind to an address.. now it does :) |
ASTERISK-01317: Yellow alarms on E1 PRI with current zaptel |
ASTERISK-01318: badly configured voicemail.conf causes segfault |
ASTERISK-01319: [patch] Fix AbsoluteTimeout extension T |
ASTERISK-01320: [patch] Add file-management applications to asterisk |
ASTERISK-01321: [patch] Small patch to Makefile |
ASTERISK-01322: Hangup during voicemail locks zap channel |
ASTERISK-01323: Codec conversion between ulaw and alaw using IAX causes audio gremlins/ lots of clicks |
ASTERISK-01324: PLEASE HELP! Transfered call gets dropped 1 sec later |
ASTERISK-01325: Proxy-Authoriation response spread over multiple lines is not accepted |
ASTERISK-01326: Asterisk Core dump with chan_h323 |
ASTERISK-01327: Using SIPDtmfMode(inband) when calling voicemail causes crash |
ASTERISK-01328: [Request] Can Hangup have a "cause" parameter |
ASTERISK-01329: [Request] Control over calling number for int/ext calls |
ASTERISK-01330: CDR-ODBC and MS-Sql 2000 no entries are made, but CDR-ODBC reports success |
ASTERISK-01331: Dlink DVG1120s fails to register after 1 week operation |
ASTERISK-01332: [patch] musiconhold url streaming bug |
ASTERISK-01333: [patch] g723 codec seems to calc ast_frame.samples incorrectly |
ASTERISK-01334: [patch] manager.c uses portno= instead of port= config in manager.conf |
ASTERISK-01335: [patch] Enabled the speed dial features for phones using chan_skinny |
ASTERISK-01336: Asterisk CallerID cannot handle all E.164 formats |
ASTERISK-01337: [patch] Calls in spool/outgoing cannot call numbers including # |
ASTERISK-01338: [request] Numeric sounds and internationalization |
ASTERISK-01339: pri show span 1 desplays wrong information for EuroISDN |
ASTERISK-01340: [patch] SayNumber-s in Danish |
ASTERISK-01341: [patch] Call Ref: debug output is confusing. |
ASTERISK-01342: Zap Channels on Adtran 750 Channel Bank Stuck in Rsrvd State |
ASTERISK-01343: [patch] Internal Manager Listener |
ASTERISK-01344: [patch] Improved Call Limit Control For Dial |
ASTERISK-01345: LibPri dies after about 90k calls |
ASTERISK-01346: qualify and change of host |
ASTERISK-01347: [patch] Potential problems with SEEK_FORCECUR |
ASTERISK-01348: [patch] Improve debug output. Add a bit more info. |
ASTERISK-01349: [request] app Directory(): Better functionality |
ASTERISK-01350: isdn pri setup/release problem. |
ASTERISK-01351: *CLI> remove extension ? causes server to freeze instantly |
ASTERISK-01352: [patch] Voicemail broadcasts |
ASTERISK-01353: chan_skinny.so (undefined symbol: ast_pickup_call) |
ASTERISK-01354: Asterisk send connect instead of alert message received from h323 channel side (chan_oh323.so) |
ASTERISK-01355: zaptel mmx option broken on kernel 2.6.5 |
ASTERISK-01356: [patch] Optional support of MAD MP3 player |
ASTERISK-01357: [patch] Prevent double unlock of channel's lock in ast_read |
ASTERISK-01358: G726 as compatible codec - recognition by various SIP devices |
ASTERISK-01359: asterisk is unable to get * variables from an AGI Script called within the hangup extension |
ASTERISK-01360: [patch] stdint.h nonexistent on FreeBSD |
ASTERISK-01361: zaptel no longer compiles on opteron (amd64) |
ASTERISK-01362: [patch] Allow you to enter specific channel in Zapscan |
ASTERISK-01363: [patch] French prompts and numbers in say.c |
ASTERISK-01364: timestamps not updated properly when using IAX2 trunking. |
ASTERISK-01365: IAX jitter buffer reorder problem [diagnosis/description] |
ASTERISK-01366: chan_oss doesnt compile in current cvs |
ASTERISK-01367: ztcfg causes loss of clocking on E1. |
ASTERISK-01368: Strange behavior with successive Dials |
ASTERISK-01369: Bad echo of DIS when sending fax |
ASTERISK-01370: missing queue announce wav causes call drop |
ASTERISK-01371: Asterisk Seg Faults consistently |
ASTERISK-01372: chan_alsa hangup fails. |
ASTERISK-01373: [PATCH] to add alarm counters to wct4xxp.c |
ASTERISK-01374: forward voicemail to extension crashes asterisk service with RH9 |
ASTERISK-01375: [request] Support F-States. |
ASTERISK-01376: CONFIG_ZAPATA_NET fails to build on Linux 2.6 |
ASTERISK-01377: chan_h323.c does not compile with cvs -head 04/08/04 |
ASTERISK-01378: ISDN Overlap-mode incoming digits doubled. |
ASTERISK-01379: Asterisk-addon's - cdr_addon_mysql fails to compile. |
ASTERISK-01380: YELLOW alarms on TE410P t1/d4/ami |
ASTERISK-01381: [PATCH] Implement SETUP T303 retransmissions. |
ASTERISK-01382: Compile of wcfxs.c (CVS version 1.45) fails with gcc 2.95.4 20011002 (Debian prerelease) |
ASTERISK-01383: [patch] Australian Indications |
ASTERISK-01384: wct4xxp.ko / zaptel.ko can't be unloaded on kernel 2.6.5 |
ASTERISK-01385: SIP channels remain open with Elesign ESC1710 and ethernet loss |
ASTERISK-01386: zaptel "channel is dead" flood |
ASTERISK-01387: SIP with videosupport _allways_ adds video to SDP |
ASTERISK-01388: asterisk dies (remote) under heavy load |
ASTERISK-01389: [patch] Length of audio file format |
ASTERISK-01390: ztcfg does not automatically run on boot if /usr is not on root filesystem (debian specific) |
ASTERISK-01391: IAXy user doesn't hear ringing tone despite ,r option in dial |
ASTERISK-01392: [request] for a ztdummy that uses usb-ohci (and works with SMP) |
ASTERISK-01393: when running asterisk as non root user, sip fails to bind to 0.0.0.0 |
ASTERISK-01394: [request] alaw codec for iaxy |
ASTERISK-01395: Error in Makefile |
ASTERISK-01396: FreeBSD 4.9 and MacOS X lack recursive lock initializers |
ASTERISK-01397: __mgcp_xmit |
ASTERISK-01398: [patch] app_queue and cdr updates |
ASTERISK-01399: [PATCH] chan_h323.c needs the recent change to dtmf duration that was made elsewhere |
ASTERISK-01400: Ringing doesn't work with the 'r' option in app_dial |
ASTERISK-01401: Crash on channel cleanup |
ASTERISK-01402: Horrible sound quality SIP -> IAX2 |
ASTERISK-01403: GET DATA fails with no input, while app_read and menus work |
ASTERISK-01404: Call hung up during initial VM prompts does not report 'Hangup' event to astman |
ASTERISK-01405: [patch] make app_queue log AgentCallBackLogin |
ASTERISK-01406: [patch] Modules are not "reloaded" in same order as they are loaded. |
ASTERISK-01407: [patch] losing digits while pri astchannel is already up |
ASTERISK-01408: Zap channel stuck bridged to itself |
ASTERISK-01409: [patch] Provide CVS Tag revision in ASTERISKVERSION (or head if none) |
ASTERISK-01410: Cannot compile Zaptel CVS STABLE |
ASTERISK-01411: [patch] Incoming SIP calls from SIP provider get wack channel names |
ASTERISK-01412: [patch] SubString is deprecated, and we should say so |
ASTERISK-01413: with current cvs head _1NXX beats _18[00|88|77|66|55] even when in separate included contexts |
ASTERISK-01414: [design] General i18n patch for say.c |
ASTERISK-01415: Some SIP clients due to poor firmware do not terminate correctly with * resulting in very long almost silent calls |
ASTERISK-01416: [patch] iax2 show peers always seems to show 3ms delay |
ASTERISK-01417: [patch] Zonedata update for Australia data |
ASTERISK-01418: [request] backports for 1.0-stable |
ASTERISK-01419: Dial should have an IP connection timeout - separate from timeout. |
ASTERISK-01420: #ifdef [patch] #ifdef ZAPATA_PRI missing from chan_zap.c ? |
ASTERISK-01421: [patch] SIP BYE isn't authorized due to wrong username being sent |
ASTERISK-01422: [patch] SMS() text message sending/receiving to ETSI ES 201 912 |
ASTERISK-01423: If both allow=ulaw and allow=adpcm are in iax.conf and iaxy only allows adpcm I get one way audio and warnings from channel.c |
ASTERISK-01424: README.festival has inaccuracy |
ASTERISK-01425: CLI 'show modules' crash Asterisk /FreeBSD |
ASTERISK-01426: [patch] say.h has wrong def of ast_say_number |
ASTERISK-01427: [patch] enum patch to get a caller name stored in TXT dns record |
ASTERISK-01428: [patch] ast_say_digit_str aborts when non-digit characters are in the string |
ASTERISK-01429: During conversation of two sip agents on put other on hold, other can take it off hold by pressing hold. |
ASTERISK-01430: VMail.cgi fix for multi context, and just to make it WORK |
ASTERISK-01431: Re: 1110 - asterisk -rx "sip show peers" or "iax2 show channels" etc does not give full response. |
ASTERISK-01432: New Patch for the lastest CVS version |
ASTERISK-01433: [patch] Proper initialization of Asterisk mutexes in debug mode |
ASTERISK-01434: cdr_clf.c for Common Log Format cdr records (webalize them) |
ASTERISK-01435: Asterisk coredump |
ASTERISK-01436: [request] Add RBS to DACS |
ASTERISK-01437: chan_zap doesn't check for multiple / ambiguous matches on PRI overlap dialling |
ASTERISK-01438: [patch] for SIP phones asterisk shows itself as caller ID when caller ID is actually unknown/undefined |
ASTERISK-01439: Asterisk Crashed |
ASTERISK-01440: passwords can't be used with SIP behind NAT as SIP clients have trouble registering with secret and auth defined |
ASTERISK-01441: [patch] Uniqueids missing in Link and Unlink manager events |
ASTERISK-01442: [patch] Change location of digits from relative to absolute paths |
ASTERISK-01443: On incoming call IAX authenticates again last peer of list only |
ASTERISK-01444: Digest buffer too small? |
ASTERISK-01445: [patch] misc nitpicks... |
ASTERISK-01446: [patch] show file formats |
ASTERISK-01447: Any load on H.323 seg faults |
ASTERISK-01448: [patch] Localize all find user, find peer, update peer calls to single functions, add mysql lookup ability to sip_devicestate() |
ASTERISK-01449: pbx_builtin_goto has a minor memory leak |
ASTERISK-01450: [patch] CLI help error: help logger rotate |
ASTERISK-01451: Reentrant gethostbyname |
ASTERISK-01452: SIP registration to down machine causes Asterisk to open to many files |
ASTERISK-01453: [patch] directory needs configuration options |
ASTERISK-01454: [patch] 'sip show peers' additional parameters |
ASTERISK-01455: [patch] indications.conf for Russia (RU) |
ASTERISK-01456: timestamps slip going sip to sip with latest cvs-head. |
ASTERISK-01457: [patch] agi sometimes gets fdopen error |
ASTERISK-01458: New Zealand Indications.conf |
ASTERISK-01459: SIP registration fails with Bad Request 400 until "reload" is executed |
ASTERISK-01460: [request + patches] iax.conf doesn't have language param |
ASTERISK-01461: iax2 crash |
ASTERISK-01462: Very vad Audio on SIP -> CAPI with some phones... |
ASTERISK-01463: [patch] Clean up SQL queries |
ASTERISK-01464: Asterisk not receiving any sip or iax calls |
ASTERISK-01465: Red Hat init.d script should not restart on reload |
ASTERISK-01466: [patch] New retrieve_extensions_from_sql.pl |
ASTERISK-01467: [patch] Advanced Voicemail Features (carryover from 156) |
ASTERISK-01468: asterisk manager interface: action id not included in response to command action |
ASTERISK-01469: [design] Internationalization of app_voicemail |
ASTERISK-01470: [patch] Update app_sql_postgres.c documentation |
ASTERISK-01471: overlapdial=yes affects DTMF |
ASTERISK-01472: [patch] chan_mgcp.c does not process retrans_pkt/mgcpsock_read not called during reload command |
ASTERISK-01473: [patch] chan_mgcp.c does not process retrans_pkt/mgcpsock_read not called during reload command |
ASTERISK-01474: Compile fails on OS X (10.3.3) |
ASTERISK-01475: app_txtcidname fails to compile |
ASTERISK-01476: Record-Route not being properly handled |
ASTERISK-01477: AutoLogoff does not work for AgentLogin. |
ASTERISK-01478: Asterisk not dealing with DTMF very well from some SIP phones |
ASTERISK-01479: host= in SIP client config not always recognized |
ASTERISK-01480: [patch] chan_zap will not compile in current CVS without libpri |
ASTERISK-01481: [patch] remove buffer from mpg123 so that it will never suck all cpu time |
ASTERISK-01482: stateful echotraining |
ASTERISK-01483: Channel properties will be setted after "ast_setstate" in several channel modules |
ASTERISK-01484: X100P coming "off hook" pegs CPU |
ASTERISK-01485: [patch] app_directory |
ASTERISK-01486: ever increasing jitter buffer with iax2 -head |
ASTERISK-01487: Voicemail w/MySQL support ignores context |
ASTERISK-01488: [PATCH] Add "accountcode" support to MySQL based IAX-FRIENDS |
ASTERISK-01489: [request] Roll the Queue log into the general logging utilities |
ASTERISK-01490: [patch] H323 doesn't compile on non-linux |
ASTERISK-01491: executing reload in -vvvvgc sometimes causes "use STOP NOW" message |
ASTERISK-01492: executing reload in -vvvvgc sometimes causes "use STOP NOW" message |
ASTERISK-01493: executing reload in -vvvvgc sometimes causes "use STOP NOW" message |
ASTERISK-01494: Key Bounce (with SIP INFO messages) |
ASTERISK-01495: sip_mysql_friends and mysql_friends configuration in Makefile |
ASTERISK-01496: [patch] Italian sound files complete set |
ASTERISK-01497: [patch] voicemail does not properly populate global options down to users with <4 commas |
ASTERISK-01498: [patch] - design suggestion: Codec names unification |
ASTERISK-01499: No ringing indication on outgoing calls |
ASTERISK-01500: Asterisk crash after success h323 call |
ASTERISK-01501: [patch] Extend app_meetme to be able to return if conference has only 1 conferee |
ASTERISK-01502: [PATCH] add HasVoicemail application |
ASTERISK-01503: VM email refers to "message (number 0)" |
ASTERISK-01504: TDM400P with FXO Modules doesn't always initialize |
ASTERISK-01505: [patch] Centralise IAX Friends and SIP friends into one table and add the ability to enable and disable users |
ASTERISK-01506: [request] Improved Directory usability |
ASTERISK-01507: PlayBack and Background applications garble their sounds |
ASTERISK-01508: * crashes on hangup UNLESS in console mode |
ASTERISK-01509: Poor sound quality |
ASTERISK-01510: [request] holdandannounce app |
ASTERISK-01511: TDM400 FXO fxs_ks disconnect supervision |
ASTERISK-01512: [request] Improved Voicemail Notification options |
ASTERISK-01513: voicemail.conf.sample has a wrap error |
ASTERISK-01514: [patch] say.c tweaks to docs & finish enabling German syntax |
ASTERISK-01515: Authentication with FWD is broken |
ASTERISK-01516: Invite Authentication fails |
ASTERISK-01517: [patch] Add requested color codes for ASTERISK_PROMPT |
ASTERISK-01518: [pre-1.0]? [patch] MeetMe, Hold everyone until Admin Login |
ASTERISK-01519: [patch] i18n support for Time & Date functions in say.c |
ASTERISK-01520: [patch] SIP Register response without Expires: header not parsing Contact: and Dynamic expiry guard time calculation |
ASTERISK-01521: [request] Full support of SIP 484 "incomplete address" |
ASTERISK-01522: [post-1.0][request] Add voice and call control encryption to IAX |
ASTERISK-01523: [patch] Config option to allow queues with no members to be joined |
ASTERISK-01524: [patch] a new channel group for calls aggregation |
ASTERISK-01525: MGCP Compatibility Issues |
ASTERISK-01526: [patch] Add swedish support, fix german support |
ASTERISK-01527: [request] add autologoff feature to AgentLogin or AddQueueMember |
ASTERISK-01528: [patch] Change strlen() to !ast_strlen_zero() |
ASTERISK-01529: [request] add "agent logoff" command to CLI |
ASTERISK-01530: agi makefile makes agi-sphinx-test, but doesn't clean it |
ASTERISK-01531: ast_expr.y: another update |
ASTERISK-01532: [patch] Change strlen() to !ast_strlen_zero() |
ASTERISK-01533: [patch] Fix Swedish support, add gender to French |
ASTERISK-01534: Compile fails on OS X (10.3.3) |
ASTERISK-01535: [patch] Change strlen() to !ast_strlen_zero() |
ASTERISK-01536: [patch] utils.h -- ast_strlen_zero() potential crash protection |
ASTERISK-01537: [patch] README.variables needs an update |
ASTERISK-01538: configuration in other than asterisk configuration format |
ASTERISK-01539: DISA Dialtone is somewhat broken. |
ASTERISK-01540: [patch] Fix language and musiconhold in chan_sip.c |
ASTERISK-01541: [patch] Improved support for SIP INFO |
ASTERISK-01542: H323 protocol error (roundtrip delay) |
ASTERISK-01543: [Patch] Would like 'r' option for app_queue to work |
ASTERISK-01544: [patch] Asterisk stops |
ASTERISK-01545: app_sms (sms_handleincoming: Unknown message type 01) |
ASTERISK-01546: [CAPI] Asterisk Crashes on Incoming Call |
ASTERISK-01547: [patch] say.c - Gender support for Spanish, Mexican to use Spanish syntax |
ASTERISK-01548: [patch] Add realm config to chan_sip |
ASTERISK-01549: possible iax2 authentication regression from stable to head |
ASTERISK-01550: [patch] 'sip show subscriptions' |
ASTERISK-01551: iax2_show_users() has b0rked strncpy(3) calls. (easy fix, no patch included) |
ASTERISK-01552: music on hold plays extremely loud and distorted |
ASTERISK-01553: Seg 11 Crash of Asterisk |
ASTERISK-01554: [patch] Add "sip show peer <name>" and minor change to "sip show peers" |
ASTERISK-01555: SIP - IAX2 - SIP Voice Quality |
ASTERISK-01556: VoiceTronix Interface driver |
ASTERISK-01557: Asterisk crashes with following syntax error in extensions.conf: exten => 1000,1,Dial(IAX2/60.254.236.xxx${EXTEN}) |
ASTERISK-01558: dbget does not jump to pri n+101 for keys not found |
ASTERISK-01559: Compiler warning in astman.c |
ASTERISK-01560: notransfer=yes does not appear to work |
ASTERISK-01561: Core dump with app_queue.c |
ASTERISK-01562: [design] Outbound Proxy Implementation in chan_sip |
ASTERISK-01563: Improper copy of musicclass string in recent update of chan_sip.c |
ASTERISK-01564: [patch] Spanish syntax for say_date_with_format |
ASTERISK-01565: A few more strlen() optimizations in chan_sip |
ASTERISK-01566: [patch] Use Asterisk ACL routines for local network matching |
ASTERISK-01567: [patch] Show callgroups and pickupgroups in "sip show peer <name>" |
ASTERISK-01568: [patch] sip show peers formatting fix |
ASTERISK-01569: [patch] GroupCheck will segfault if a channel has a null GROUP |
ASTERISK-01570: usleep in rtp.c of 500ms causes audible blip before call is bridged. |
ASTERISK-01571: Asterisk crashes when originating call to unknow extension (-stable) |
ASTERISK-01572: [patch] Remove jitter from "sip show channels" |
ASTERISK-01573: Grandstream BT-101 SIP Phones can no longer enter DTMF Digits |
ASTERISK-01574: [patch] SIP useragent and small fixes to "sip show channel" |
ASTERISK-01575: app_senddtmf.c needs to include <asterisk/app.h> |
ASTERISK-01576: CDR Error log on call transfer |
ASTERISK-01577: Need to add ; to queues.conf.sample |
ASTERISK-01578: mime header tips for Linuz Zaurus Mailer (SL-A,SL-B,SL-C) |
ASTERISK-01579: [patch] fix charset of mail header to be compatible with chineese |
ASTERISK-01580: Taiwan indications.conf |
ASTERISK-01581: [patch] zonedata for Taiwan |
ASTERISK-01582: chan_modem_i4l.c MAX_WRITE_SIZE needs to be increaed to 1280 |
ASTERISK-01583: [patch] MIME boundary text can trigger delays in email from voicemail |
ASTERISK-01584: [patch] Add explanations of functions in chan_sip.c |
ASTERISK-01585: Strange things with sip show channels |
ASTERISK-01586: [patch] voicemail shows "Re-Recording the message" when it's being recorded for the first time. This patch fixes that. |
ASTERISK-01587: [patch] Improvements to German saynumber() |
ASTERISK-01588: asterisk coredump |
ASTERISK-01589: ast_gethostbyname interprets a peername with all numbers as a integers based IP address |
ASTERISK-01590: [patch] manager_event and run_externnotify don't use context |
ASTERISK-01591: CVS STABLE V1.0 apparently still has broken Digest Authentication code |
ASTERISK-01592: Oneway audio from SIP devices with latest CVS after 2004-05-08 |
ASTERISK-01593: BT-102 DTMF not accepted in AGI |
ASTERISK-01594: CHAN_PHONE: Caller ID display is faulty |
ASTERISK-01595: Alsa can only connect in one direction |
ASTERISK-01596: [patch] Add Taiwanese support for saynumber() & saydate() |
ASTERISK-01597: Pressing # on a sip channel sends tone and gives error in rtp.c |
ASTERISK-01598: SIP-connections stay open in some cases |
ASTERISK-01599: Incorrect header for app_transfer.c |
ASTERISK-01600: Bringing hdlc0 interface down and back up causes kernel panic |
ASTERISK-01601: [patch] Compile error in say.c |
ASTERISK-01602: spool outgoing failed call |
ASTERISK-01603: Add -C <configfile> option to command line help (-h) output |
ASTERISK-01604: Lose Dial tone on TDM400P randomly |
ASTERISK-01605: [patch] Extend app_meetme with several features |
ASTERISK-01606: Patch that adds events reporting meetme join/leave. |
ASTERISK-01607: FreeBSD 5.2.1 doesn't have CIRCQ |
ASTERISK-01608: [patch] remove auth= variable in config |
ASTERISK-01609: PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP undefined on FreeBSD 5.2.1 |
ASTERISK-01610: [Patch]Polish say_number |
ASTERISK-01611: [Patch] Basic code to remove hardcoded ENTER and LEAVE signal |
ASTERISK-01612: [patch] IMPORTANT FIX FOR DEBIAN USERS + Simplification of say.c |
ASTERISK-01613: [Design] MeetMe enter/leave signal |
ASTERISK-01614: [patch] Improvement of chan_sip.c debugging |
ASTERISK-01615: fixed return code for gethostbyname_r for FreeBSD |
ASTERISK-01616: AF_INET needed for sending UDP on FreeBSD 5.2.1 |
ASTERISK-01617: [patch] AF_INET needed for sending UDP on FreeBSD 5.2.1 |
ASTERISK-01618: B-channels not restarting properly due to slow response from PBX |
ASTERISK-01619: mysql-vm-routines.h: no "default" context |
ASTERISK-01620: Receive packets with DTMF from D-Link MGCP gateway |
ASTERISK-01621: 2 TDM04B w/4 fxo modules each |
ASTERISK-01622: [patch] add libsrtp |
ASTERISK-01623: [patch] minor timestamp cleaning changes; jitter buffer knobs. |
ASTERISK-01624: [patch] libiax2: combined sync with iaxclient: jitterbuffer, acking, PING response, and more. |
ASTERISK-01625: None ring back - h323. |
ASTERISK-01626: Outbound DTMF Fails with Telica - RFC2833 Sends DTMF Packets with Identical Timestamp and Sequence Numbers |
ASTERISK-01627: Outbound DTMF Fails with Telica - RFC2833 Sends DTMF Packets with Identical Timestamp and Sequence Numbers |
ASTERISK-01628: [Patch] Correct Copyright welcome message |
ASTERISK-01629: Can't compile say.c (CVS-HEAD) |
ASTERISK-01630: coredump when bridging chan h323 to iax2 |
ASTERISK-01631: [request] Add timestamp to sip debug output |
ASTERISK-01632: spurious DTMF digit detection terminates call |
ASTERISK-01633: TDM400 FXO module (maybe all FXO) |
ASTERISK-01634: app_app.c: timezone offset on freebsd |
ASTERISK-01635: FreeBSD lack oss (like Darwin) |
ASTERISK-01636: [PATCH] Tones for Chile |
ASTERISK-01637: __ast_request_and_dial: Don't know what to do with control frame 14 |
ASTERISK-01638: [Patch] Allow vmail.cgi to be specified a default context |
ASTERISK-01639: D Channels die under outbound load |
ASTERISK-01640: Zaptel cannot compile on latest RHEL Kernels |
ASTERISK-01641: Timestamps in some RTP streams do not increment |
ASTERISK-01642: [patch] Add language support to chan_local.c |
ASTERISK-01643: Allow interface restriction for RTP |
ASTERISK-01644: Allow localisation of timestamps in logger module and use ISO 8601 default |
ASTERISK-01645: busy tone |
ASTERISK-01646: Add secondary ENUM lookup to enum.conf |
ASTERISK-01647: meetme.conf needs final newline |
ASTERISK-01648: Audio Glitching - to much junk cached... |
ASTERISK-01649: Cisco 7960 poor sound quality with cvs |
ASTERISK-01650: Asterisk DEADLOCKED on latest cvs |
ASTERISK-01651: Audio Glitching - to much stuff cached... |
ASTERISK-01652: Audio Glitching. To much stuff cached. |
ASTERISK-01653: ALSA Device or resource busy error with CVS HEAD |
ASTERISK-01654: Incorrect calling party number parsing for debuging purpose |
ASTERISK-01655: Call hangs up after a while when "#" transfer to SIPtone IP phone |
ASTERISK-01656: Dial l command no longer has the c or confirmation option |
ASTERISK-01657: ADSI configuration in VM is incorrect |
ASTERISK-01658: [patch] Congestion() followed by Hangup() should send PRI cause code 34 |
ASTERISK-01659: [PATCH] Ncurses/gcc problems without termcap. One Liner fix |
ASTERISK-01660: [patch] Add ability to turn on/off timestamp playback on voicemail playback for user. |
ASTERISK-01661: vm-from-extension and vm-from-phonenumber say "message from" |
ASTERISK-01662: Loop when HD fills up |
ASTERISK-01663: "3 for advanced options" on voicemail is bad audio quality |
ASTERISK-01664: Libpri does not support CodeSet 6 (struct q931_ie is invalid) |
ASTERISK-01665: Blocked CallerIDs appear as from 'asterisk' not as 'unavailable' |
ASTERISK-01666: Zero out of voicemail problem |
ASTERISK-01667: Correct IAX message count reporting to return a valid message count value. |
ASTERISK-01668: Duplicate codec entries in SDP |
ASTERISK-01669: [request] libpri disconnect cause passed over iax2 |
ASTERISK-01670: [patch] chan_agent.c: adds new options for CLI Functions AgentLogin() and AgentCallbackLogin() |
ASTERISK-01671: HANGUPCAUSE is 0 |
ASTERISK-01672: Asterisk deadlock - IAX ok, but not receiving SIP |
ASTERISK-01673: [patch] Lag responses cannot go through jitter buffer |
ASTERISK-01674: [patch] Voicemail removal script deletes unheard voicemails |
ASTERISK-01675: patch to compile libpri on FreeBSD |
ASTERISK-01676: qualify=yes in iax.conf causes incorrect Status |
ASTERISK-01677: When making outgoing call over PRI, Asterisk sends PROCEEDING and PROGRESS messages together with SETUP |
ASTERISK-01678: Q931_PROG_CALL_NOT_E2E_ISDN value invalid |
ASTERISK-01679: [Patch] MailboxExists doesn't work |
ASTERISK-01680: [patch] Add doc and some formatting to app_meetme (cvs head) |
ASTERISK-01681: Zaptel crashes or hangs system on FC2 2.6 Kernel |
ASTERISK-01682: AddQueueMember command doesn't support wrapuptime |
ASTERISK-01683: AddQueueMember command doesn't support penalty flags |
ASTERISK-01684: [patch] Make res_monitor use SPOOL variable instead of hardcoded one |
ASTERISK-01685: Default ADSI Script (asterisk.adsi) Has Incorrect Security Codes... |
ASTERISK-01686: PRI: Read on 39 failed: Unknown error 500 (PRI resets) |
ASTERISK-01687: main Makefile doesn't work with MacOS 10.2 |
ASTERISK-01688: md5.c and aesopt.h don't handle MacOS X |
ASTERISK-01689: ANSWER not recognized with trunking (possibly seqno issue) |
ASTERISK-01690: latest chan_h323 doesnt seem to like my gnugk / openphone |
ASTERISK-01691: chan_local ignores context |
ASTERISK-01692: [request] Fastforward and rewind in a playback application |
ASTERISK-01693: zaptel modules cleanup for 2.6.x |
ASTERISK-01694: [patch] ztdummy driver for 2.6.x |
ASTERISK-01695: [patch] Caller ID support for UK (BT POTS lines) |
ASTERISK-01696: [patch] ENUM support for ifax:mailto |
ASTERISK-01697: show g729 on new g729 codec displays wrong num of used licenses |
ASTERISK-01698: Call scheduled for auto destruction in 15 seconds if type=peer |
ASTERISK-01699: [patch] Add realm authentication to SIP |
ASTERISK-01700: chan_323 crash on connect |
ASTERISK-01701: Re-invite codec negotiation bug |
ASTERISK-01702: SECURITY: remotely exploitable heap overflow in Asterisk |
ASTERISK-01703: Queue application does not properly update lastcall time |
ASTERISK-01704: Codecs list table (h.323 show codecs) increments after each connection... |
ASTERISK-01705: [patch] Doc for meetme & ast_strlen_zero |
ASTERISK-01706: Asterisk crash |
ASTERISK-01707: Application that says phrases acording to given pattern |
ASTERISK-01708: PRI related |
ASTERISK-01709: [patch] ACL copying routines |
ASTERISK-01710: Failed SIP md5 REGISTER in two simultaneous requests |
ASTERISK-01711: SIP channels not hanging up |
ASTERISK-01712: Compile chan_oss and app_intercom on FreeBSD |
ASTERISK-01713: [Patch] add a exit context to voicemail for "0" and "*" options |
ASTERISK-01714: Incoming calls from FWD no longer work since CVS HEAD 2004/05/24 |
ASTERISK-01715: Unknown A-number (4823522) is displayed (stable cvs version) when no A-number is present |
ASTERISK-01716: Type of Number |
ASTERISK-01717: Using variables in voicemail.conf / fromstring |
ASTERISK-01718: iax2/gsm timestamps not consistent at 20ms |
ASTERISK-01719: Chan Zap gets funky after MeetMe conf |
ASTERISK-01720: TDM400P 4 FXO Modules busydetect does not work |
ASTERISK-01721: rtptimeout is not right - scary stuff!!!!! |
ASTERISK-01722: When using g729 as primary codec, media description line in the SDP lists Codec type 18 twice |
ASTERISK-01723: [request] TDM40B (FXO modules) support for complex impedances |
ASTERISK-01724: Incoming calls from FWD no longer work since CVS HEAD 2004/05/24 |
ASTERISK-01725: [request] Support battery polarity reversal |
ASTERISK-01726: TDM400 FXO module detect when Monitor Phone connect to line as ring |
ASTERISK-01727: support for variable codecs frame size |
ASTERISK-01728: SIP authentication failure is back |
ASTERISK-01729: eliminate compile time warning about unused LOADAVG on FreeBSD |
ASTERISK-01730: serialize dns resolver search |
ASTERISK-01731: elmiinate compile time warning about mkdir() on FreeBDS` |
ASTERISK-01732: Liniking cdr_scv and chan_oss on FreeBSD |
ASTERISK-01733: Variables with space do not work it uses the first word as the wole variable. |
ASTERISK-01734: VoicemailMain2 does not respond in anticipated manner |
ASTERISK-01735: voicemail with USE_MYSQL_VM_INTERFACE=1 does ignore context definition |
ASTERISK-01736: Macros do not clear argument variables when calling other macros |
ASTERISK-01737: In NoOp and System , is still converted to | |
ASTERISK-01738: SIP handset Transfer fails and Multi-line fails |
ASTERISK-01739: Terrible sound quality |
ASTERISK-01740: [patch] Modularized external query API modules |
ASTERISK-01741: [patch] meetme - make the announcement fd into a proper channel |
ASTERISK-01742: Timing out on a recording will not play acknowledgment message. |
ASTERISK-01743: Using Wildcard X101P FXO Card after a couple of minutes causes kernel panic on 64bit Athlon |
ASTERISK-01744: app_read.c doesn't return entered digits unless # pressed. |
ASTERISK-01745: ASTERISK deadlock |
ASTERISK-01746: Asterisk Crashes with Seg Fault - may be associated with moh |
ASTERISK-01747: MOH. server freezes when loading 10 chars after loading music_onhold |
ASTERISK-01748: [patch] app_meetme.c crashes when removing non-head or tail user from non-first conference |
ASTERISK-01749: TDM 4-port FXO First port doesn't work |
ASTERISK-01750: [patch] Adds events and queue_log messages for Callback Agents |
ASTERISK-01751: Groups defined in zapata.conf not working after recent update |
ASTERISK-01752: Zap does take calls automatically |
ASTERISK-01753: [post-1.0 discussion] [patch] Implementation of labels in the dialplan |
ASTERISK-01754: [patch] Problems with string quotes,escaping and regular expression in extensions.conf |
ASTERISK-01755: SIP/SDP Payload Type problems |
ASTERISK-01756: zaptel and oss dependencies for FreeBSD |
ASTERISK-01757: [Request] The hability to do options while liscening to voicemail |
ASTERISK-01758: [patch] New app: Ademco Contact ID alarm receiver |
ASTERISK-01759: [patch] libpri extended debugging |
ASTERISK-01760: [patch] PRI notifications support |
ASTERISK-01761: [patch] Transmission of character set in Display IE is not regulated by Q.931 |
ASTERISK-01762: timestamp issue with codec conversion and sound quality |
ASTERISK-01763: Quicknet Internet Phonejack crashes asterisk on incomming calls |
ASTERISK-01764: extensions.conf |
ASTERISK-01765: [patch] Fixes crash when joining monitor files |
ASTERISK-01766: [patch] Add an integrated user conference feature |
ASTERISK-01767: Message waiting doesnt work when using MYSQL for sipfriends. |
ASTERISK-01768: Problem between ChanIsAvail and app_dial |
ASTERISK-01769: [patch] change order of example code in Voicemail.conf |
ASTERISK-01770: app_voicemail.c error as of today's cvs update |
ASTERISK-01771: asterisk fails to start dchanel wrongly set to 24 on E1 |
ASTERISK-01772: [patch] Add DBget and DBput to manager interface |
ASTERISK-01773: Transaction Mismatching in Register |
ASTERISK-01774: Zero out of voice mail problems in 1.0-stable |
ASTERISK-01775: asterisk crashes |
ASTERISK-01776: kill() undefined at compile time for app_zapras.c on FreeBSD |
ASTERISK-01777: ast_log when asterisk -cdv segfaults |
ASTERISK-01778: Called Voicemail system will trigger fax bridge |
ASTERISK-01779: MD5 portability to 64-bit |
ASTERISK-01780: [patch] srvlookup defaults to off - non RFC compliant. |
ASTERISK-01781: Call drops during socket bind in rtp.c |
ASTERISK-01782: QuickNet Internet Phonejack Hangs up on ULAW Codec |
ASTERISK-01783: [patch] error building chan_h323 after mutex relative changes |
ASTERISK-01784: PRI busy state confusion |
ASTERISK-01785: #define of AST_MUTEX_KIND in lock.h needs diffferent #ifdef |
ASTERISK-01786: Zap channel lockout for meetme initiated calls after hangup |
ASTERISK-01787: format_vox.c has stab functions that should be filled with code |
ASTERISK-01788: Asterisk in deadlock |
ASTERISK-01789: asterisk crash when usecallerid = yes |
ASTERISK-01790: [patch] DB implementation of SIP "type=user" |
ASTERISK-01791: [patch] Merged changes from FreeBSD Asterisk port |
ASTERISK-01792: dsp.c won't compile with OLD_DSP_ROUTINES |
ASTERISK-01793: Weird behavior when you do early-dial and IAX |
ASTERISK-01794: Weird behavior when you do early-dial and IAX |
ASTERISK-01795: Asterisk doesnt run on an Opteron (compiles OK though). |
ASTERISK-01796: [patch] Priority Queues |
ASTERISK-01797: Quick Net Internet Phonejack Goes Bezerk |
ASTERISK-01798: CDR MySQL add-on build breaks in Asterisk-1.0 |
ASTERISK-01799: Kernel crash during connecting / disconnecting form the conferencing |
ASTERISK-01800: [patch] pulse dialing for chan_zap |
ASTERISK-01801: extensions.conf breaks when there are commas inside of command arguments |
ASTERISK-01802: Too short a buffer in extensions.conf parser |
ASTERISK-01803: [post-1.0][patch] Improvements in app_voicemail.c |
ASTERISK-01804: [post-1.0][patch] Adds the ability to copy a channel variable from any channel to the current channel |
ASTERISK-01805: OpenBSD compile fixes |
ASTERISK-01806: [patch] Overlap dialing not accepted after (premature) proceeding on PRI |
ASTERISK-01807: No dial tone on a TE405P in PRI mode |
ASTERISK-01808: Problems handling incoming alerting and incoming connected on a pri |
ASTERISK-01809: [patch] Swedish indications |
ASTERISK-01810: [patch] Alaw support im app_meetme |
ASTERISK-01811: System panics - I think its to do with ZAP. |
ASTERISK-01812: [request] change name of functions: load_module, unload_module |
ASTERISK-01813: [request] load_module, ans similar should be named ast_load_module or other |
ASTERISK-01814: T1 PRI chronic Red Alarms, but no alarm on zttool utility |
ASTERISK-01815: [patch] remote hold notification support |
ASTERISK-01816: [patch] Added support for Remote-Party-ID |
ASTERISK-01817: [patch] Change cli output |
ASTERISK-01818: refer fails with compact headers |
ASTERISK-01819: [patch] zaptel Makefile assumes existence of $(INSTALL_PREFIX)/usr/include/linux |
ASTERISK-01820: Cannot use I4l + GSM unless this patch is done |
ASTERISK-01821: Prevent guess of IAX2 username and pass. |
ASTERISK-01822: Leak in editline |
ASTERISK-01823: Periodic reset of idle PRI B-channels affect non idle as well |
ASTERISK-01824: [PATCH] Patch that repairs the alsa-chan complete |
ASTERISK-01825: CLI commands always show last iax2.conf entry when entries have same host |
ASTERISK-01826: CLI commands always show last iax2.conf entry when entries have same host |
ASTERISK-01827: iax2.conf type=user should be able to drive codec acceptance |
ASTERISK-01828: init_logger_chain() called twice in a row in logger.c |
ASTERISK-01829: wcfxo.c needs yet another PCI device id... |
ASTERISK-01830: [PATCH] Two leaks in logger.c |
ASTERISK-01831: VoiceMailMain behaves unexpectedly |
ASTERISK-01832: Asterisk not sending "Sending Complete" on EuroISDN overlap span |
ASTERISK-01833: [patch] dynamic configuration in app_queue.c |
ASTERISK-01834: [post-1.0][patch] MSSQL CDR Backend |
ASTERISK-01835: [patch] fixes for EOF conditions in app_sql_postgres.c |
ASTERISK-01836: Asterisk sometimes hangup call on IAX2 transfer |
ASTERISK-01837: SIP Registration with Softswitch header information missing |
ASTERISK-01838: localtime.c won't compile with MALLOC_DEBUG |
ASTERISK-01839: IAXY using adpcm |
ASTERISK-01840: [patch] More voicemail fixes |
ASTERISK-01841: Calls transfered from Agents don't hear MOH |
ASTERISK-01842: [patch] Document the fact useragent can be changed in sip.conf |
ASTERISK-01843: CDR update is delayed if the called party hangsup |
ASTERISK-01844: MeetMe causes PBX to lock up (g729 possibly involved) |
ASTERISK-01845: MGCP channel sends RQNT, CRCX commands out of order |
ASTERISK-01846: app_queue does not behave as expected with empty queue. |
ASTERISK-01847: Something changed-- double quotes are now removed before eval-- exprs affected |
ASTERISK-01848: Manager originate command to IAX channel fails |
ASTERISK-01849: AddQueueMember with penalty |
ASTERISK-01850: Race between incoming overlap digits in app_dial and setup_ack |
ASTERISK-01851: Overlap digits not stored in the cdr |
ASTERISK-01852: Useful enhancement to chan_zap would be to fill in subaddresses etc |
ASTERISK-01853: meetme marked users (x option) broken |
ASTERISK-01854: Asterisk ringback tones on SIP channels |
ASTERISK-01855: MWI-cann't receive correct format notify sip message |
ASTERISK-01856: Quoted expressions in extension logic broken again |
ASTERISK-01857: asterisk coredumps |
ASTERISK-01858: asterisk coredumps |
ASTERISK-01859: System lockups. |
ASTERISK-01860: [patch] ast_safe_system() always returns -1 |
ASTERISK-01861: Asterisk crashing after releasing clone lock |
ASTERISK-01862: [patch] app_directory does nothing if last matching extension is skipped by user and last entry is last extension |
ASTERISK-01863: Outgoing SIP Proxy authentication failes due to missing credentials |
ASTERISK-01864: No audio with different IPs for audio & signalling endpoints |
ASTERISK-01865: no output from "h.323 show codecs" |
ASTERISK-01866: Asterisk crashes on latest cvs |
ASTERISK-01867: [patch] Confirming entry in Directory restarts Directory |
ASTERISK-01868: Busy message on unavailable - app_dial |
ASTERISK-01869: Every debug message from IAX2 that has 2 or less characters has ug appeneded |
ASTERISK-01870: answer while overlap dial ISDN |
ASTERISK-01871: [request] Short key for [1-9] in extensions.conf would be useful |
ASTERISK-01872: Asterisk deadlocks - gdb output attached |
ASTERISK-01873: tdm fxo does not sense broken pstn line (no battery) |
ASTERISK-01874: tdm fxo senses pstn line disturbances as ringing |
ASTERISK-01875: Asterisk crashed - back trace attached |
ASTERISK-01876: Core dump in ast_cli when doing a sip show peer |
ASTERISK-01877: tdm fxo echo when calling from sip phone |
ASTERISK-01878: [request] Option to escape queue by pressing # or * |
ASTERISK-01879: fax problems with V.17 |
ASTERISK-01880: [patch] File specified in ast_say_date_with_format should be relative |
ASTERISK-01881: Manager API "Command" does not produce full response.as expected |
ASTERISK-01882: parking call confuses/hangs server after cvs update |
ASTERISK-01883: FATAL: Error inserting zaptel |
ASTERISK-01884: wrong channel state for outgoing calls on Zap |
ASTERISK-01885: Voicemail Help Menu repeats option 3 |
ASTERISK-01886: Create Makefile.inc, enable more warnings, resolve warnings. |
ASTERISK-01887: Makefile needs libcrypto for FreeBSD |
ASTERISK-01888: [patch] Add Unique ID to Asterisk Manager Event JOIN |
ASTERISK-01889: Siemens optiPoint 400 sends '%23' instead of '#' |
ASTERISK-01890: Cancel branch after final response. |
ASTERISK-01891: Mixmonitor stopped storing |
ASTERISK-01892: [patch] cmd line flag to reconnect remote asterisk clients when disconnected |
ASTERISK-01893: [patch] add channel parameter to manager status command |
ASTERISK-01894: [patch] add manager commands to toggle chan_zap DND |
ASTERISK-01895: [patch] character class in : operator regexp terminates expression |
ASTERISK-01896: Can't unload/reload codec_g729a.so |
ASTERISK-01897: No way to take incoming call if another in progress |
ASTERISK-01898: Manager originate command from SIP channel to ZAP channel fails |
ASTERISK-01899: Compile error in pbx.c |
ASTERISK-01900: wcfxs.c fails to build on 2.6, missing interrupt.h? |
ASTERISK-01901: Request - Complex impedance matching for TDM400 FXS modules |
ASTERISK-01902: Add AT&T NSF support to libpri and asterisk |
ASTERISK-01903: IAX connections without username from dynamic host allowed with only "secret" |
ASTERISK-01904: Add ability to specify peercontext in iax.conf |
ASTERISK-01905: [patch] astman won't build |
ASTERISK-01906: Zaptel variable type mismatch |
ASTERISK-01907: zaptel cvs fails to compile on kernel 2.6.7 |
ASTERISK-01908: Zaptel wcfxs stop answering call on TDM400 FXO |
ASTERISK-01909: [patch] Possible fix for empty option |
ASTERISK-01910: Make RFC3581 support optional |
ASTERISK-01911: Asterisk coredump |
ASTERISK-01912: Remove astman compilation warnings |
ASTERISK-01913: [patch] Add agents.conf custom_beep parameter |
ASTERISK-01914: pbx_config 1.43 removes quotes ' that it shouldn't |
ASTERISK-01915: Current CVS branch does not compile on OS X (10.3) |
ASTERISK-01916: Enhancement to app_queue to give the seconds for the hold time |
ASTERISK-01917: rtp.c gets frequent EAGAIN on rtcp packets |
ASTERISK-01918: 'a' and 'o' extensions do not work |
ASTERISK-01919: inet_ntoa is not thread-safe, replace with inet_ntop |
ASTERISK-01920: Typo in name of file to stream |
ASTERISK-01921: [patch] Fix asterisk.spec so it can build a rpm package for current asterisk CVS |
ASTERISK-01922: [patch] Adds 'noanswer' option to Read and Record applications |
ASTERISK-01923: [patch] document INVALID_EXTEN in README.variables |
ASTERISK-01924: [patch] utils.c will not compile on FreeBSD |
ASTERISK-01925: cvs head 20040629-14:45 |
ASTERISK-01926: [patch] Code cleanups for pbx_wilcalu.c |
ASTERISK-01927: zaptel fails to build on 2.4.18 kernel |
ASTERISK-01928: build of chan_iax2.c failed when USE_MYSQL_FRIENDS defined |
ASTERISK-01929: Compiling against the full 2.6 sourcecode is not required |
ASTERISK-01930: Unable to reduce iax2 registration times |
ASTERISK-01931: [patch] Use INET_ADDRSTRLEN to define all iabuf[] strings for use with ast_inet_ntoa() calls |
ASTERISK-01932: [patch] Stop app_voicemail.c from sending email if no email defined for user |
ASTERISK-01933: Music on hold randomly skips around |
ASTERISK-01934: Current CVS Breaks GrandStream and possibly other SIP devices |
ASTERISK-01935: Poort Quality Issues using IAX2 and Cisco 7960 SIP Phones |
ASTERISK-01936: crackle and high-pitched squeal for call waiting |
ASTERISK-01937: [patch] missing context when callling app_hasvoicemail |
ASTERISK-01938: asterisk picks up immediate when call comes from zap channel |
ASTERISK-01939: Problem with transferring callers from the queue |
ASTERISK-01940: app_txtcidname.so fail to load due to undefined symbol |
ASTERISK-01941: app_disa does not handle near end hangup |
ASTERISK-01942: Truncated output on -x 'database show' for large databases |
ASTERISK-01943: [request] CLI error reporting |
ASTERISK-01944: [request] variables available in emailtitle |
ASTERISK-01945: CVS-HEAD 7-3-04 has a Typo in the default iax.conf file |
ASTERISK-01946: App_Voicemail externnotify always returns 1 |
ASTERISK-01947: [request] new manager API Command - 'database' |
ASTERISK-01948: [patch] Allow Manager API Command to return big amount of output data +++ |
ASTERISK-01949: SIP 30X Reply (promicious redirect) fails with authentication on INVITE for Nikotel |
ASTERISK-01950: [patch] New features for LineJack phone driver |
ASTERISK-01951: call lost after "Attempting native bridge " |
ASTERISK-01952: [patch] app_voicemail.c bad strncpy() and snprintf() call |
ASTERISK-01953: No RTP audio when transcoding SIP <->H323 |
ASTERISK-01954: Answer Confirmation doesn't work on PRI |
ASTERISK-01955: Calls hanging up when answered |
ASTERISK-01956: chan_h323 error (maybe human? ;) ) |
ASTERISK-01957: app_voicemail does not tell user password bad with VoicemailMain(mailbox@context) format |
ASTERISK-01958: TDM FXO hook status error |
ASTERISK-01959: [patch] Make voicemail act more normal like, and jump to the next message afer del/save if configured in voicemail.conf to do so |
ASTERISK-01960: Get chan_sip.c : Failed to grab lock trying.... ---------after one SIP to SIP call |
ASTERISK-01961: [PATCH] cdr_sqlite |
ASTERISK-01962: *8 call pickup from sip to sip sometimes callee keeps ringing |
ASTERISK-01963: [patch] Make iax2-provision.c compile under FreeBSD 4.9 |
ASTERISK-01964: [patch] Rtp media stream doesn't honor bindaddr if the addr is an alias on an interface |
ASTERISK-01965: [patch][src-audit] chan_sip.c: Fixed possible buffer overruns, general cleanup |
ASTERISK-01966: Call parking does not work via SIP "Transfer" button |
ASTERISK-01967: Bug in sample extensions.conf breaks drop to voice mail |
ASTERISK-01968: Bad output in ast_monitor for context of voicemail in MessageWaiting report |
ASTERISK-01969: udev does'nt create appropriate device files into /dev/ |
ASTERISK-01970: Get the correct expire-time from REGISTER response |
ASTERISK-01971: [patch] app_voicemail.c patch in #1971 undid part of #1977 |
ASTERISK-01972: HANGUPCAUSE not being set |
ASTERISK-01973: [request] PRI cause codes reported as AST_CAUSE_FAILURE |
ASTERISK-01974: [patch] Allow Busy to return immediately, with flag |
ASTERISK-01975: X100P Ring/Off-hook in strange state 6 |
ASTERISK-01976: adds incominglimit and outgoinglimit to mysql user |
ASTERISK-01977: unload chan_iax2.so press tab... segfault. & res_config doesnt work with chan_iax2 |
ASTERISK-01978: [patches][src-audit] app.c, asterisk.c, callerid.c, cdr.c, channel.c, cli.c, config.c, db.c |
ASTERISK-01979: We're ZAP/2-1, not Zap/5-1 |
ASTERISK-01980: first second is truncated if call is established through an ip gateway |
ASTERISK-01981: [patch] Fix parsing of .call files |
ASTERISK-01982: [patch] Debian init script |
ASTERISK-01983: Horrible echo issues with X1ooP |
ASTERISK-01984: [patch] allow the transfer keys from app_dial's 't' and 'T' and hangup key 'H' to be configured via features.conf |
ASTERISK-01985: [patches][src-audit] cli.c, dlfcn.c bufferoverrun, malloc checks |
ASTERISK-01986: Read/Write Cisco Remote-Party-ID |
ASTERISK-01987: [patch] vm_intro_cz |
ASTERISK-01988: Pause before last digit dialed on zap FXO channel |
ASTERISK-01989: Unreliable Dial on TDM400P FXO Channels |
ASTERISK-01990: [request + patch] Please implement RP-AS : Ring Puls Alert Signal. (CallerID) |
ASTERISK-01991: sip.conf.sample explains how user and peer works incorrectly |
ASTERISK-01992: chan_local does not inherit channel variables or account codes |
ASTERISK-01993: WARNING[196621]: chan_iax2.c:2515 iax2_send: Out of trunk data space on call number 16385, dropping |
ASTERISK-01994: ztd-local driver |
ASTERISK-01995: VM recorded from pstn-TDM-Zap is ~-10db (low volume) |
ASTERISK-01996: [request] Add channel-specific VM gain parameter |
ASTERISK-01997: sip.conf type=peer not distinguishing hostname with type=user |
ASTERISK-01998: [patch] Directory() usable from VoiceMail - Forward |
ASTERISK-01999: CDR billsec/duration should reflect queue holdtime |